Pw44,Before you do anything, turn the firewall off and try all your registrations and see if they work. The process of elimination so to speak. Then

Do a "sip set debug on" then do a "sip show peers". Post the results here or send to my email I will try to see what's happening with SPA.

Also, I see that you have listed sip.conf and sip_additional_conf. Have you actually modified these files and added what you needed on the new machine with 10.04? On newer versions of * sip.conf and others like it are rewritten when you reload *, older versions didn't do that. Check to make sure that your sip.con doesn't say at the beginning like "to modify another conf file as this file is overwritten when reloading * . I will read through you conf files posted tonight when I get free and see if I can tell what's going on.

I am planning tonight on seeing if it is just possible to set LMCE up as a client to a external * machine and be done with trying to make * in LMCE work.

microbrain

Hi Microbrain,thx for yor offer in analyze the debug.As told, i do have: 2 sccp devices (cisco 7970), extensions 201 and 203, two media directors, extensions 200 and 202 and a spa-3102, where line 1 is set as extension 204. spa-3102 is connected in the inside network (192.168.80.0), with not access from the external network. Is only a pstn to voip gateway to enable the use of the old pstn.As sip trunks, i have voipcheap, but sound only one way, and the pstn line from spa-3102 does not registers on asterisk.I did enable nat, and the needed parameters at the database tables, disabled the firewall, but no way to have it working.sip set debug on gave the following results, and i hope that some can see what i'm not being able to.Best regards and thx again.Paulo

The first thing you need to do is get all the "401" & "403" response codes fixed.

I'm seeing that your VoIp provider is not rejecting(401 code) then accepting. Educate me again on some things:

How does you machine communicate with the "outside" world? From LMCE to a router, to a DSL/Cable modem and are you using a static IP? If dynamic IP have you registered with someone like dyndns.org or are you just dependant on your Internet Service Provider?

On the spa, if I am correct I'm seeing 401 & 403s, you need to check that the user name & password are the same as what LMCE has entered in its database. It could be that it is and the database is not storing the passwords the same. I have had this issue before in the past, a corrupt database.....

I'm also seeing that the spa is trying to communicate on port 5061, is this the only port in the spa that is assigned, if so you need to have it at beginning at 5060 and end with blank or at least 5061.I'm assuming that it is set for 5061, asterisk normally starts with 5060....

Had to look your post over pretty quick as I have to leave for church but will check it again when I return but in the passing time check for all your 401 & 403 and try to correct them, as I say it's a authentication issue. I have a website that will give you the sip responses at: http://en.wikipedia.org/wiki/List_of_SIP_response_codes and you can always go to asterisk web site for additional info.

The first thing you need to do is get all the "401" & "403" response codes fixed.

I'm seeing that your VoIp provider is not rejecting(401 code) then accepting. Educate me again on some things:

How does you machine communicate with the "outside" world? From LMCE to a router, to a DSL/Cable modem and are you using a static IP? If dynamic IP have you registered with someone like dyndns.org or are you just dependant on your Internet Service Provider?

On the spa, if I am correct I'm seeing 401 & 403s, you need to check that the user name & password are the same as what LMCE has entered in its database. It could be that it is and the database is not storing the passwords the same. I have had this issue before in the past, a corrupt database.....

I'm also seeing that the spa is trying to communicate on port 5061, is this the only port in the spa that is assigned, if so you need to have it at beginning at 5060 and end with blank or at least 5061.I'm assuming that it is set for 5061, asterisk normally starts with 5060....

The spa3102 is connected in the internal network, as the log shows (192.168.80.30).The spa configuration has two parts: pstn and line 1.Line 1 is defined as extension, and registers as a sip phone.The pstn is defined with port 5061 and user and password were checked and are the same. The whole configuration of the spa3102 is the same as the wiki, http://wiki.linuxmce.org/index.php/Linksys_SPA3102, including username and password Subscriber Settings

Display Name - Unknown Caller (this is what is displayed when a call comes in without caller ID info) UserID - spa3102 (this is what you set when creating the Phone Line in the LinuxMCE webmin) Password - lmce (this is what you set when creating the Phone Line in the LinuxMCE webmin)

In the 8.10 release, it worked well, with the new release, it does not authenticates, and i'm not finding out the reason, as userid, password, port and settings are the same.

Quote

Had to look your post over pretty quick as I have to leave for church but will check it again when I return but in the passing time check for all your 401 & 403 and try to correct them, as I say it's a authentication issue. I have a website that will give you the sip responses at: http://en.wikipedia.org/wiki/List_of_SIP_response_codes and you can always go to asterisk web site for additional info.

Well, something new to report:having 2 sccp extensions and 1 sip extension, the results are:calling from outside to my voipcheap trunk, sip extension rings but not the sccp (yes, all are configured to ring)calling from outside to my sipgate trunk, sip extension rings, but not the sccp (yes, all are configured to ring).Answering the call from the sip extension, vice both ways (perfect).Calling from any sccp extension to the sip, voice one way only.Anyone with a mixed environment (sccp and sip extensions)?And i did find out that the /usr/share/asterisk/sounds/pluto are wrong (no voicemail)......

we don't under the way LMCE has it setup now. Not all VoIp providers have a "Standard" of connection. It would be nice if they did as it would make life easier. That's where something like FreePBX comes in so that you can modify necessary information. Just like under "Phones" we don't have the ability to choose the codecs, the Outbound CID Number Alias, and a host of other options.

If LMCE could find it in their design to at least code the phone line page & phones page to resemble the same as is under FreePBX it would be great.

I, under the circumstances, don't feel I can help you any further as I see the problems being all within LMCE. It will take a rewrite of some code to fix.

Microbrain,you are right. On the previous version, 8.10, with freepbx, i got all running as you said.With the new one, i'm not finding where and how to make it..... and maybe that's the problem: my ignorance and lack of documentation Anyway, thx again for trying to help.Best regards,Paulo

mcefan, if you find it useful to add a link to this thread, why didn't you

Because I had no clue where that was! Unfortunately, people assume others know. I for one, don't usually work with these things, so, this is a learning experience for me, and in the process, I made up my mind to try to lower the entry curve for others. Not everyone interested in the project is a developer, and I really believe that a larger user base will help the project a great deal, and that's my goal. The entry level learning curve is ridiculously steep, so, every time you see me asking or commenting, I'm not necessarily thinking about myself.

I don't know what trac is...but since the subject came up, I looked on the main site and found the link "Tracker" under "Developer", which takes me here. Seeing that I am not a developer, I would not have ventured under that heading.I suggest we add a simple paragraph to the wiki that will point people to it, with a simple explanation of when to do so, and links to the most important pages there.

The most frustrating is that is almost impossible to get help and support.... no one knows and the few that knows are silent... in the asterisk forum, etc... If there was any decent documentation... but this is also a no show.my spa3102 is there, as nmap shows:

As I said earlier I think it's going to be a bug within LMCE and it's db. Please run the following command from a command line on the LMCE machine:

sip set debug ip 192.168.0.3 (put in your IP address of you LMCE machine that the SPA is trying to communicate on)

The post here so I can see what it is doing. Since there is no peer found running sip show peer won't show what's going on between the SPA and LMCE, using the "IP" will.

Asterisk is not finding the peer name you are trying to register with the SPA. There is either a user name or password not matching up with what Asterisk is looking for based on what Asterisk is pulling from it's db, normally Asterisk pulls this info from it's AstDB.

Also while your at it, copy and paste your "sip.conf" file here, you can find it on the LMCE server /etc/asterisk/sip.conf, oh wait a minute, there is no sip.conf on the LMCE machine I have under etc/asterisk..... Either it is located somewhere else and I'm not aware of or found yet, or, it's been renamed to something else, or, maybe someone forgot to include it in the build....

I will be tied up till after ten tonight so I won't be able to get right back to you, but I will look at it tonight. Stop pulling your hair out, I'm sure there is a simple answer to it all.

sip debug for asp3102 ata ip:Please note that the spa3102 is serving fro two purposes:1 - pstn line as a pots trunk - no way no register.2 - line 1 as a sip extension 204 - this one registers and works.

Also while your at it, copy and paste your "sip.conf" file here, you can find it on the LMCE server /etc/asterisk/sip.conf, oh wait a minute, there is no sip.conf on the LMCE machine I have under etc/asterisk..... Either it is located somewhere else and I'm not aware of or found yet, or, it's been renamed to something else, or, maybe someone forgot to include it in the build....

Your problem is within the authentication of your PSTN line., but you know that already.

Three things I would look at: (make note of any changes you make)

1: Did you set the spa for dhcp or assign it a dynamic ip? If you have it as dhcp, change it to static and try that then change the #2 next, if it still don't work put it back.

2: in the spa under "Proxy and Registration" is it set for yes, if so try to set it to off and then check if it works. If it is set to on it tries to tell LMCE what your spa address is, and since you already assigned an ip you don't need to register.

3: as much as I hate to say this, go back to LMCE 8.10 because the current 10.04 will not work without the ability to setup this box with something like (here it comes, wait, wait,) FREEPBX...... and don't feel too bad as you won't be the only one having to do it.

I myself was going to try and set LMCE up to be an extension of my asterisk box, but after studying how LMCE communicates with asterisk I can't do that either nor can I make the asterisk box work as a in/out trunk to the LMCE because of the changes they made in how the LMCE deals with asterisk. I'm still not sure if I fully understand why the powers to be had to change up asterisk/lmce. If it truly was an issue (as I have read on some other post) that too many people were breaking LMCE using FreePbx I would have thought it would have been easier to just make FreePBX an add-on and not offered help when they screwed it up, just my opinion.